Embodiments of the present invention relate to the provision of telephony services. More particularly, embodiments of the present invention are directed to an automated system for generating dial plans for telephone switching equipment.
With the passage of the Telecommunications Act (“the Act”) of 1996, an incumbent local exchange carrier (ILEC), the regulated entity that owns and administers an existing access network, must provide to a requesting telecommunications carrier (hereinafter generally referred to as “Competitive Local Exchange Carrier” or “CLEC”) nondiscriminatory access to network elements on an unbundled basis and allow CLECs to combine such network elements in order to provide telecommunications services. ILECs also have a duty to provide to CLECs interconnection with their network for the transmission and routing of telephone exchange service and exchange access. The interconnection contemplated by the Act provides nondiscriminatory access or interconnection to such services or information as are necessary to allow the requesting CLEC to implement local dialing parity, including nondiscriminatory access to telephone numbers, operator service, directory assistance, and directory listing, with no unreasonable dialing delays.
The provisions of the Act have demonstrated a need for competing exchange carriers to be interconnected so that customers can seamlessly receive calls that originate on another carrier's network and place calls that terminate on another's carrier's network without performing additional activities, such as dialing extra digits, etc.
The increasing popularity of high speed data (HSD) over cable and the emergence of voice over IP (VoIP) technology as a viable alternative to the public switched network (PSTN) has provided cable operators an opportunity to offer a full range of VoIP-based telephony services, offering custom features and advanced intelligent network services that rival the ILECs.
A cable operator desiring to provide VoIP services directly to customers either acts as a CLEC or partners with a CLEC. In this capacity, the cable operator manages not only the HSD network but a telephone switch (variously referred to as a softswitch, media gateway controller, call agent, or a gatekeeper) that manages calls originating from, and terminating with, subscribers. In addition to managing VoIP calls on the network, the telephone switch manages the routing of calls to and from the PSTN.
FIG. 3 illustrates the network components of a PacketCable-compliant system. Referring to FIG. 3, a managed IP network 325 is shown connected to a Fiber 320A and to the PSTN 350 through a trunking media gateway 340. (VoIP over cable under the PacketCable standard differs from VoIP services provided over the Internet in that the service is not delivered over the public Internet but over a managed IP network.)
The access network 315A connects a subscriber's cable modem (CM) 305A and multimedia (or media) terminal adapter (MTA) 310A to the CMTS 320A at the cable headend. The MTA 310A handles voice compression, packetization, security, and call signaling to support a standard telephone 300A and a fax machine (not shown) through an RJ-11 connector. An MTA 310A may be designed to be either a separate standalone device or to be embedded within the CM 305A (an EMTA). The MTA 310A and the CM 305A are assigned separate media access control (MAC) and IP addresses even if the elements are integrated into a single device.
The CMTS 320A uses the Data Over Cable Service Interface Specification (DOCSIS) 1.1 protocol (also issued by Cable Television Laboratories, Inc.) on the access network 315A to manage access network resources for PacketCable services. Access network 315A resources are first reserved when service is requested, then committed when service is delivered, and finally released when the service has completed. Softswitch 330 manages and maintains a call state for VoIP services and controls the MTA 310A.
The PacketCable Specification specifies various devices for signaling, call control, and connectivity. A signaling gateway (SG) provides SS7 signaling to the PSTN. A media gateway controller (MGC) provides signaling to the media gateways, such as the cable modem termination system (CMTS) and the media gateway (MG). The MG provides connectivity to the PSTN. A call management system (CMS) provides call control within the network, primarily call setup and tear down.
FIG. 3 illustrates a number of these components. By way of illustration, softswitch 330 provides signaling gateway (SG), media gateway control (MGC), and CM functionality. The softswitch functions as the center for call control. It provides call features, maintains call state, and controls every call setup and teardown. Additionally, the softswitch has full control over the residential gateways in the network and routes calls to the appropriate destinations, such as to other MTAs or traditional phones. This functionality is provided by two subsystems, a call agent (CA) 332 and an element management system 336 (EMS).
CA 332 provides call logic and call control functions, maintaining call state for every call in the network, and performs the CMS, MGC, and SG functions specified in the PacketCable Specification. The CA 332 includes service logic for supplementary services, such as, Caller ID, Call Waiting, and also interacts with application servers to supply services that are not directly hosted on the call agent. The CA 332 participates in signaling and device control flows originating, terminating or forwarding messages. There are numerous relevant protocols in use on the call agent, including: SIP, MGCP/NCS, SS7, and AIN. Call agents also produce details of each call to support billing and reconciliation.
The EMS 336 is a mediation device between a network management system (NMS) (not illustrated) and a call agent 332. The EMS 336 facilitates the provisioning, administration, reporting, and billing features of the softswitch 330.
On-net to off-net calls are handled by trunking media gateway 340. The trunking media gateway 340 connects directly to a CLEC switch (class 4 or 5) and provides a gateway between a digital phone network and the PSTN 350.
The CA 332 interacts with the MTA 310A and other functional entities to establish calls by using the device control functions provided by network-based call signaling (NCS) Protocol. A gate controller (GC) (not illustrated) maintains a policy database and interacts with the MTA 310A and the gate function of the CMTS 320A to coordinate the quality-of-service (QoS) authorization, admission control, and related channel attributes configuration.
Several different types of signaling are deployed in a digital phone network. All of the signaling protocols work together to push a call through the network.
Media gateway control protocols are used by the media gateway controller (MGC) to control media gateways in a digital phone network. Media Gateway Control Protocol (MGCP) also known as RFC2705/RFC 3435 is a Network Working Group specification. Megaco, also known as H.248, was co-developed by the ITU and the Megaco Working Group. The Session Initiation Protocol (SIP) was developed to control various SIP-enabled end devices. If a media gateway is SIP-enabled, then SIP becomes a media gateway control protocol. The Common Open Policy Service (COPS) is used between the softswitch and CMTS for gate-control communication.
At the subscriber level, network-based call signaling (NCS) is used. NCS is a simplified version of MGCP developed specifically to control embedded digital phone client devices in a PacketCable environment.
The network services signaling protocols allow the softswitch to communicate with other networks. The primary protocols used for this purpose are Signaling System 7 (SS7) as illustrated in FIG. 3 (345), SIP and/or CMSS. SS7 is the network service signaling protocol that supports softswitch communications with the PSTN. SIP is designed to control SIP enabled endpoints. Any device in the network can be a SIP endpoint, even another softswitch. Various telecommunications service providers are implementing SIP soft switches as an alternative to SS7.
The PacketCable specifications uses SIP with extensions that support local and custom local area signaling services (CLASS) services generally supported in a telephony network. This protocol is referred to as Call Management Server Signaling (CMSS).
IP communications use the Real-time Transport Protocol (RTP) and the Real-time Transport Control Protocol (RTCP) to negotiate the connections between IP devices. RTP provides end-to-end network transport functions for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services. RTP does not address resource reservation and does not guarantee quality-of service for real-time services. The data transport is augmented by the RTCP control protocol that allows monitoring of the data delivery in a manner scalable to large multicast networks while providing minimal control and identification functionality. RTP and RTCP are designed to be independent of the underlying transport and network layers.
Bearer connectivity refers to the connection of the physical interfaces that actually carry the voice traffic. As previously noted, signaling traffic is carried separately from voice traffic. In a true digital phone network, all data (voice and data) traffic would be carried over an IP-based infrastructure. However, since today's telephone network is made up of time division multiplexed (TDM) connections, there must be a mechanism to pass traffic to the TDM network as well as the IP network.
From a cable operator's perspective, there are two types of telephone calls. On-Net Calls are calls that are carried entirely on the digital phone network and do not interface with the PSTN. Off-Net calls are calls that must traverse the PSTN to reach their destination. Off-net calls are carried through the trunking media gateway to the PSTN. The trunking media gateway translates the IP voice packets into the TDM format and sends them out to the PSTN. The connection between the trunking media gateway and the PSTN is through inter-machine trunks or IMTs. IMTs are generally high bandwidth (DS3) connections that connect directly to the CLEC Class 5 switch.
One of the components of the telephone switch is a dial plan. A dial plan provides the basic functionality to meet customer requirements for calling and called number analysis. The rule set used by a switch to determine, for every combination of calling party and dialed digits, how to route the call (e.g. international, long distance, toll, local, on-net, off-net or “vacant code”) is called a dial plan. Calls that are determined to be “on-net” are routed according to the “on-net” components of the dial plan.
The dial plan specifies what types of service requests can be accommodated by a telephone switch based on the call setup information, the calling number, the called number, and the routes, trunk groups, and trunks (individual circuits) available to the telephone switch. The dial plan determines how each and every call that comes into telephone switch is processed.
As calls arrive in the call processing system, the originating and terminating telephone numbers along with other data are evaluated to determine what call processing actions will be taken. Various actions can be implemented, such as call acceptance or rejection based on the calling or called number, what call control instructions are used, whether the call is on-net or off-net, and, for off-net calls, which egress trunk (or circuit) is ultimately selected to carry the call.
A typical dial plan is a single file organized as a set of tables, each in a different section. The different sections are linked lists of values to be used after number analysis is completed.
The North American Numbering Plan (NANP), on which all U.S. telephone numbers are currently based, is an integrated telephone numbering plan serving nineteen North American countries that share its resources. The NANP is administered by a North American Numbering Plan Administrator (NANPA). NANP numbers are ten-digit numbers consisting of a three-digit Numbering Plan Area (NPA) code, commonly called an area code, followed by a three digit central office code, also known as exchanges or prefixes, and a four-digit local, or line, number. The format is usually represented as NPA-NXX-XXXX where N is any digit from 2 through 9 and X is any digit from 0 through 9. Each area code is divided geographically into rate centers, geographic areas that are billed as if they were a single location.
Rate centers are a holdover from the days when a separate switch was needed for almost every community, and so calls between communities—even within a LATA—were “rated” as toll calls. When each rate center corresponded to a discrete switch, every rate center/switch needed its own NXX. Currently, a single Class 5 typically serves multiple rate centers. By way of illustration, in Maine, 133 rate centers are served by 15 host switches and 13 smaller switches.
Despite the technological realities, the rate center concept still determines the numbering scheme of the wired telephone system. Currently, two different rate centers served out of the same Class 5 switch must each have their own NXX, no matter how few lines each represents. The reason is that the prefix designates the local switching equipment. When a call is placed to NPA-NXX-XXXX, the telephone network examines the area code and prefix in order to route the call to the correct switch. The far-end switch then selects the correct line based on the line number. Because carriers do not typically share a local switch, each prefix must designate both the wire center, the location of the switch, and the carrier.
The numbering plan is not static. Area codes are added (either by splitting an existing area into two NPAs or overlaying a second area code on a single NPA) to accommodate number shortages and because of competitive demands. Alternatively, rate centers can be consolidated to make more efficient use of the blocks of numbers that are issued under a single area code or added. Such changes have a significant impact on the way telephone switches are configured. Dial-plans, in particular, are huge tables of routing rules that undergo significant churn in response to these changes to the NANP.
By way of illustration, an 878 NPA was added in Pennsylvania to serve the same area served by the 412 and 724 area codes. With the overlay, a new dial plan was implemented requiring that after a “permissive period” all calls within the 412, 724 and 878 areas were to be dialed as 10-digits, and all calls outside 412, 724, and 878 were to be dialed as 1+10-digits. While the change was made to a region in Pennsylvania, all telephone switching systems were required to be modified to reflect the new NPA in the dial plan and to require 10-digit and 1+ten-digit dialing. Had the 878 NPA been added by splitting one of the other two NPAs, the impact on dial plans would have been more significant.
A telephone switch also generates billing information based on the service types defined in the dial plan. Off-net service types typically fall within one of the following groups:                “Offshore” NPAs—calls terminating at an NPA outside a rate center LATA and outside a single-rate long distance service plan.        Long Distance NPAs—calls terminating at an NPA outside a rate center LATA that are within a single-rate long distance service plan.        Intra-LATA Toll—calls terminating at an NPA within a rate center LATA that are outside the local calling area and are billed at local toll rate.        Intra-LATA Local—calls terminating at an NPA within a rate center LATA that are within the local calling area that are not subject to toll charges.        Inter-LATA Local—calls terminating at an NPA outside a rate center LATA that are within the local calling area that are not subject to toll charges.        
The service types are established by industry representatives and reflected in reports issued by the NANPA. LATAs are defined by NANPA. But local calling areas are actually defined by each local service provider's local service tariffs. Typically, service providers mimic the ILEC/RBOC tariffs with regard to local calling areas, since the end users are accustomed to the ILEC's tariffs.
A telephone switch dial plan must be modified to accommodate changes in the NANP as, for example, changes in the NPA and changes in the rate center. Otherwise, a call may not be completed and/or a call may not be properly billed. By way of illustration, assume that the NXX code 995 is established within the 212 NPA within a LATA 132. Since calls from outside LATA 132 are routed based only upon the 3-digit NPA (assuming a ten-digit dialing area), this new NXX does not present connect problems for calls originating from outside LATA 132. However, for a call originating within a LATA, the NPA and the NXX are required to determine whether a call is local or toll. Until the telephone switch is updated to incorporate a rule for routing calls to the new NXX code of 995, a switch serving LATA 132 will not process calls originating within LATA 132 to the 212-995 exchange, and will instead route calls to a “your call cannot be completed as dialed” announcement.
What would be useful is a dial plan generator that would receive data reflecting changes in the NANP and produce instructions in a form executable by a telephone switch to implement those changes.